Update a ton of audio code to use size_t more correctly and in general reduce

webrtc/modules/audio_device/mock_audio_device_buffer.h

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_DEVICE_MOCK_AUDIO_DEVICE_BUFFER_H_
 #define WEBRTC_MODULES_AUDIO_DEVICE_MOCK_AUDIO_DEVICE_BUFFER_H_
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "webrtc/modules/audio_device/audio_device_buffer.h"
 
 namespace webrtc {
 
 class MockAudioDeviceBuffer : public AudioDeviceBuffer {
  public:
   MockAudioDeviceBuffer() {}
   virtual ~MockAudioDeviceBuffer() {}
 
-  MOCK_METHOD1(RequestPlayoutData, int32_t(uint32_t nSamples));
+  MOCK_METHOD1(RequestPlayoutData, int32_t(size_t nSamples));
   MOCK_METHOD1(GetPlayoutData, int32_t(void* audioBuffer));
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_MOCK_AUDIO_DEVICE_BUFFER_H_